Abstract. Avaya Solution & Interoperability Test Lab

Transcription

1 Avaya Solution & Interoperability Test Lab Application Notes for Avaya Aura Communication Manager 5.2.1, Avaya Aura Session Border Controller 6.0 and Avaya Aura Session Manager 6.1with Qwest iq SIP Trunk (version 6.5.7R1) Issue 1.0 Abstract These Application Notes describe the steps to configure Session Initiation Protocol (SIP) Trunking between Qwest iq SIP Trunk (version 6.5.7R1) and an Avaya SIP-enabled enterprise solution. The Avaya solution consists of Avaya Aura Communication Manager 5.2.1, Avaya Aura Session Manager 6.1 and Avaya Aura Session Border Controller 6.0 (SBC) with various Avaya endpoints. Information in these Application Notes has been obtained through DevConnect compliance testing and additional technical discussions. Testing was conducted in the Avaya Solutions and Interoperability Test Lab, utilizing Qwest iq SIP Trunk Services. 1 of 75

2 TABLE OF CONTENTS 1. Introduction Interoperability Compliance Testing Support Test Results / Known Limitations Reference Configuration Interoperability Compliance Testing Equipment and Software Validated Configure Communication Manager Licensing and Capacity System Features IP Node Names Codecs IP Network Region Signaling Group Trunk Group Calling Party Information Outbound Routing Incoming Call Handling Treatment Configure Avaya Aura Session Manager System Manager Login and Navigation Specify SIP Domain Add Location Add Adaptation Module Add SIP Entities Add Entity Links Add Routing Policies Add Dial Patterns Add/View Session Manager Avaya Aura SBC Configuration Initial Installation Network Settings Logins VPN Access SBC Confirm Installation Verify Installation Post Installation Configuration Add Additional SIP Gateways Blocked Headers Third Party Call Control Save the Configuration Qwest iq SIP Trunk Configuration General Test Approach and Test Results of 75

3 9. Verification and Troubleshooting Conclusion References Appendix A: Avaya Aura SBC Configuration File of 75

4 1. Introduction These Application Notes describe the steps to configure Session Initiation Protocol (SIP) Trunking between Qwest iq SIP Trunk (version 6.5.7R1).and an Avaya SIP-enabled enterprise solution. The Avaya solution consists of Avaya Aura Session Border Controller (AA-SBC) 6.0, Avaya Aura Session Manager 6.1, Avaya Aura Communication Manager and various Avaya endpoints. The Qwest iq SIP Trunk service referenced within these Application Notes is positioned for customers that have an IP-PBX or IP-based network equipment with SIP functionality, but need a form of IP transport and local services to complete their solution. Qwest iq SIP Trunk will enable delivery of origination and termination of local, longdistance and toll-free traffic across a single broadband connection. A SIP signaling interface will be enabled to the Customer Premises Equipment (CPE). SIP Trunk will also offer remote DID capability for a customer wishing to offer local numbers to their customers that can be aggregated in SIP format back to customer Interoperability Compliance Testing A simulated enterprise site comprised of Communication Manager, Session Manager and the AA-SBC was connected to the public Internet using a broadband connection. The enterprise site was configured to connect to the Qwest iq SIP Trunk service through the public Internet. To verify SIP trunking interoperability, the following features and functionality were covered during the interoperability compliance test: Incoming PSTN calls to various phone types. Phone types included H.323, SIP, digital, and analog telephones at the enterprise. All inbound PSTN calls were routed to the enterprise across the SIP trunk from the service provider. Outgoing PSTN calls from various phone types. Phone types included H.323, SIP, digital, and analog telephones at the enterprise. All outbound PSTN calls were routed across the SIP trunk to the service provider. Inbound and outbound PSTN calls to/from the Avaya one-x Communicator (1XC) soft phone (H.323 version). 1XC supports two modes (Road Warrior and Telecommuter). Each supported mode was tested. 1XC also supports two versions with different firmware (H.323 and SIP). Both versions of 1XC were tested. Various call types including: local, long distance, international, outbound toll-free, operator assisted calls, local directory assistance (411), emergency assistance (911), etc. Codecs G.711MU, G.729A and G.729AB. 4 of 75

5 T.38 Fax DTMF tone transmissions passed out-band RTP events as per RFC Calling number presentation and calling number restriction. Voic navigation using DTMF for inbound and outbound calls. User features such as hold and resume, transfer, and conference. Off-net call forwarding and mobility (extension to cellular). Routing inbound PSTN calls to call center agent queues. Network Call Redirection using SIP REFER for call transfer of inbound back to PSTN. Network Call Redirection of inbound call back to PSTN from Communication Manager vector Support Avaya customers may obtain documentation and support for Avaya products by visiting In the United States, (866) GO-AVAYA ( ) provides access to overall sales and service support menus. For technical support on the Qwest iq SIP Trunk services, contact Customer Service at Enter your phone number and click Speak to us now and Customer Service will call you or select the us link to send an inquiry or click Contact a rep and fill in the request information. Note: CenturyLink acquired Qwest in April Over time Qwest branded services and web sites may be renamed by CenturyLink Test Results / Known Limitations Interoperability testing of Qwest iq SIP Trunk (version 6.5.7R1) was completed with successful results for all test cases with the exception of the observations/limitations described below. No Error Indication if No Matching Codec Offered on Inbound Calls: If the Communication Manager SIP trunk is improperly configured to have no matching codec with the service provider and an inbound call is placed, the service provider only s a 488 Not Acceptable Here response and the caller will hear a fast busy after 30 seconds. Codecs are normally agreed to upon turn-up so this condition should be discovered at that time. No Error Indication if No Matching Codec Offered on Outbound Calls: If the Communication Manager SIP trunk is improperly configured to have no matching codec with the service provider and an outbound call is placed, the service provider only s a 487 Request Terminated response. The caller will hear a fast busy and the called party will hear one ring before the call is terminated. Codecs are normally agreed to upon turn-up so this condition should be discovered at that time. 5 of 75

6 No Support for G.729B: Qwest iq SIP Trunk does not support G.729B codec. Calling Party Number (PSTN transfers): The calling party number displayed on the PSTN phone is not updated to reflect the true connected party on calls that are transferred to the PSTN. After the call transfer is complete, the calling party number displays the number of the transferring party and not the actual connected party. The PSTN phone display is ultimately controlled by the PSTN provider, thus this behavior is not necessarily indicative of a limitation of the combined Avaya/Qwest SIP Trunk solution. It is listed here simply as an observation. All Trunks Busy will ring from 7 40 seconds before fast busy: When all Communication Manager trunk-group members are busy, the caller will hear ringing for anywhere from 7 seconds to 40 seconds before finally hearing a fast busy. Qwest iq SIP Trunk will send the call to the Avaya Communication Manager and it will a 500 Service Unavailable instead of a 503 Service Unavailable. This only happens on the first call in after the trunk is busy. After the first call, the Avaya Communication Manager s 403 Forbidden. The workaround for this is to upgrade to one of the following loads: CM SP9, CM SP3, CM 6.2. Use of a 503 allows for a backoff time period and a retry by Qwest. SIP Network REFER to an off-net extension is not supported: When a Communication Manager vector is programmed to redirect an inbound call to a PSTN number after the call is answered by an announcement, Qwest iq SIP Trunk will send a 202 Accepted to the REFER SIP message and the call will drop. Qwest iq SIP Trunk does not support REFER messages. Network Call Redirection: When a Communication Manager vector is programmed to redirect an inbound call to a PSTN number before answering the call in the vector, Qwest iq SIP Trunk will send an ACK to the 302 Moved Temporarily SIP message from the enterprise but will not redirect the call to the new party in the Contact header of the 302 message. The inbound call initiator hears a recording from Qwest in this failure scenario. SIP REFER with transfer (consultative or blind) is not supported in Qwest iq SIP Trunk (version 6.5.7R1): When an extension receives a call from a PSTN number and attempts to transfer (either consultative or blind) the call to another PSTN extension, the call will initially connect and then will be dropped as soon as the transfer is completed on the enterprise user s side. This is addressed in a future Qwest iq SIP Trunk release, meanwhile the work-around is to have the Network Call Redirection field to n on page 4 of the trunk group form, refer to Section of 75

7 2. Reference Configuration Figure 1 illustrates the sample configuration used for the DevConnect compliance testing. The configuration is comprised of the Avaya CPE location connected via a T1 Internet connection to the Qwest iq SIP Trunk. The Avaya CPE location simulates a customer site. At the edge of the Avaya CPE location, an SBC provides NAT functionality and SIP header manipulation. The SBC receives traffic from Qwest iq SIP Trunk on port 5060 and sends traffic to the Qwest iq SIP Trunk using destination port 5060, using the UDP protocol. Figure 1: Avaya Interoperability Test Lab Configuration 2.1. Interoperability Compliance Testing A separate trunk was created between Communication Manager and Session Manager to carry traffic to and from the service provider. This was done so that any specific trunk or codec settings required by the service provider could be applied only to this dedicated trunk and not affect other enterprise SIP traffic. This trunk carried both inbound and outbound traffic to/from the service provider. 7 of 75

8 For inbound calls, the calls flowed from the service provider to the AA-SBC then to Session Manager. Session Manager used the configured dial patterns (or regular expressions) and routing policies to determine the recipient (in this case the Communication Manager) and on which link to send the call. Once the call arrived at Communication Manager, further incoming call treatment, such as incoming digit translations and class of service restrictions could be performed. Outbound calls to the PSTN were first processed by Communication Manager for outbound feature treatment such as automatic route selection, digit manipulation and class of service restrictions. Once Communication Manager selected the proper SIP trunk, the call was routed to Session Manager. The Session Manager once again used the configured dial patterns (or regular expressions) and routing policies to determine the route to the AA-SBC. From the AA-SBC, the call was sent to the Qwest iq SIP Trunk service via the public Internet. 3. Equipment and Software Validated The following equipment and software were used for the sample configuration provided: Equipment: Software: Avaya Aura Communication Manager SP8 running on Avaya S8300c Server (R015x ) with Avaya G430 Media Gateway Avaya Aura Session Manager 6.1 SP2 running on HP DL360 Server (Build ) Avaya Aura System Manager 6.1 SP2 running on HP DL360 Server (Build ) Patch Avaya 9641G IP Telephone (H.323) Avaya one-x Deskphone Edition 6.010f Avaya 9621G IP Telephone (SIP) Avaya one-x Deskphone SIP Avaya 4625 IP Telephone (H.323) Avaya one-x Deskphone Edition 2.5 Avaya 9600 Series IP Telephone (H.323) Avaya one-x Deskphone Edition 3.102S Avaya one-x Communicator (H.323) 6.0 with SP1 ( ) Avaya 2420D Digital Telephone n/a Avaya 6210 Analog Telephone n/a Avaya Aura Session Border Controller running on Avaya S8800 Server 8 of 75

9 4. Configure Communication Manager This section describes the procedure for configuring Communication Manager for Qwest iq SIP Trunk. A SIP trunk is established between Communication Manager and Session Manager for use by signaling traffic to and from Qwest iq SIP Trunk. It is assumed the general installation of Communication Manager and Avaya G430 Media Gateway has been previously completed and is not discussed here. The Communication Manager configuration was performed using the System Access Terminal (SAT). Some screens in this section have been abridged and highlighted for brevity and clarity in presentation. Note that the IP addresses and phone numbers shown throughout these Application Notes have been edited so that the actual public IP addresses of the network elements and public PSTN numbers are not revealed Licensing and Capacity Use the display system-parameters customer-options command to verify that the Maximum Administered SIP Trunks value on Page 2 is sufficient to support the desired number of simultaneous SIP calls across all SIP trunks at the enterprise including any trunks to the service provider. The example shows that 450 SIP trunks are available and 20 are in use. The license file installed on the system controls the maximum values for these attributes. If a required feature is not enabled or there is insufficient capacity, contact an authorized Avaya sales representative to add additional capacity. display system-parameters customer-options Page 2 of 11 OPTIONAL FEATURES IP PORT CAPACITIES USED Maximum Administered H.323 Trunks: Maximum Concurrently Registered IP Stations: Maximum Administered Remote Office Trunks: 0 0 Maximum Concurrently Registered Remote Office Stations: 0 0 Maximum Concurrently Registered IP econs: 68 0 Max Concur Registered Unauthenticated H.323 Stations: Maximum Video Capable Stations: Maximum Video Capable IP Softphones: Maximum Administered SIP Trunks: Maximum Administered Ad-hoc Video Conferencing Ports: Maximum Number of DS1 Boards with Echo Cancellation: 80 0 Maximum TN2501 VAL Boards: 0 0 Maximum Media Gateway VAL Sources: 50 1 Maximum TN2602 Boards with 80 VoIP Channels: 0 0 Maximum TN2602 Boards with 320 VoIP Channels: 0 0 Maximum Number of Expanded Meet-me Conference Ports: of 75

10 4.2. System Features Use the change system-parameters feature command to set the Trunk-to-Trunk Transfer field to all to allow incoming calls from the PSTN to be transferred to another PSTN endpoint. If for security reasons, incoming calls should not be allowed to transfer back to the PSTN then leave the field set to none. change system-parameters features Page 1 of 18 FEATURE-RELATED SYSTEM PARAMETERS Self Station Display Enabled? n Trunk-to-Trunk Transfer: all Automatic Callback with Called Party Queuing? n Automatic Callback - No Answer Timeout Interval (rings): 3 Call Park Timeout Interval (minutes): 10 Off-Premises Tone Detect Timeout Interval (seconds): 20 AAR/ARS Dial Tone Required? y On Page 9, verify that a text string has been defined to replace the Calling Party Number (CPN) for restricted or unavailable calls. This text string is entered in the two fields highlighted below. The compliance test used the value of UNKNOWN for both. change system-parameters features Page 9 of 18 FEATURE-RELATED SYSTEM PARAMETERS CPN/ANI/ICLID PARAMETERS CPN/ANI/ICLID Replacement for Restricted Calls: UNKNOWN CPN/ANI/ICLID Replacement for Unavailable Calls: UNKNOWN DISPLAY TEXT Identity When Bridging: principal User Guidance Display? n Extension only label for Team button on 96xx H.323 terminals? n INTERNATIONAL CALL ROUTING PARAMETERS Local Country Code: International Access Code: ENBLOC DIALING PARAMETERS Enable Enbloc Dialing without ARS FAC? n CALLER ID ON CALL WAITING PARAMETERS Caller ID on Call Waiting Delay Timer (msec): of 75

11 4.3. IP Node Names Use the change node-names ip command to verify that node names have been previously defined for the IP addresses of the S8300 server s Processor Ethernet (procr) and for the Session Manager SM100 interface. These node names will be needed for defining the service provider signaling group in Section 5.6. change node-names ip Name IP Address ASM CMM default procr IP NODE NAMES 4.4. Codecs Use the change ip-codec-set command to define a list of codecs to use for calls between the enterprise and the service provider. For the compliance test, codecs G.729A and G.711MU were tested using ip-codec-set 2. To use these codecs, enter G.729A and G.711MU in the Audio Codec column of the table in the order of preference. Default values can be used for all other fields. Silence suppression is normally set to n and packet size is standard at 20ms. change ip-codec-set 2 Page 1 of 2 Codec Set: 2 IP Codec Set Audio Silence Frames Packet Codec Suppression Per Pkt Size(ms) 1: G.711MU n : G.729A n 2 20 On Page 2, set the Fax Mode to T.38-standard. change ip-codec-set 2 Page 2 of 2 IP Codec Set Allow Direct-IP Multimedia? n Mode Redundancy FAX t.38-standard 0 Modem pass-through 0 TDD/TTY US 3 Clear-channel n 11 of 75

12 4.5. IP Network Region Create a separate IP network region for the service provider trunk. This allows for separate codec or quality of service settings to be used (if necessary) for calls between the enterprise and the service provider versus calls within the enterprise or elsewhere. For the compliance test, IP-network-region 10 was chosen for the service provider trunk. Use the change ipnetwork-region 10 command to configure region 10 with the following parameters: Set the Authoritative Domain field to match the SIP domain of the enterprise. In this configuration, the domain name is sip.avaya.com. This name appears in the From header of SIP messages originating from this IP region. Enter a descriptive name in the Name field. Enable IP-IP Direct Audio (shuffling) to allow audio traffic to be sent directly between IP endpoints without using media resources in the Avaya Media Gateway. Set both Intra-region and Inter-region IP-IP Direct Audio to yes. This is the default setting. Shuffling can be further restricted at the trunk level on the Signaling Group form. Set the Codec Set field to the IP codec set defined in Section 4.4. Default values can be used for all other fields. change ip-network-region 10 Page 1 of 19 IP NETWORK REGION Region: 10 Location: 1 Authoritative Domain: sip.avaya.com Name: SIP Trunks MEDIA PARAMETERS Intra-region IP-IP Direct Audio: yes Codec Set: 2 Inter-region IP-IP Direct Audio: yes UDP Port Min: 2048 IP Audio Hairpinning? y UDP Port Max: 3329 DIFFSERV/TOS PARAMETERS RTCP Reporting Enabled? y Call Control PHB Value: 46 RTCP MONITOR SERVER PARAMETERS Audio PHB Value: 46 Use Default Server Parameters? y Video PHB Value: P/Q PARAMETERS Call Control 802.1p Priority: 6 Audio 802.1p Priority: 6 Video 802.1p Priority: 5 AUDIO RESOURCE RESERVATION PARAMETERS H.323 IP ENDPOINTS RSVP Enabled? n H.323 Link Bounce Recovery? y Idle Traffic Interval (sec): 20 Keep-Alive Interval (sec): 5 Keep-Alive Count: 5 12 of 75

13 On Page 3, define the IP codec set to be used for traffic between region 10 and any other regions that are in use. Enter the desired IP codec set in the codec set column of the row with destination regions (dst rgn) 1 and 2. Default values may be used for all other fields. The example below shows the settings used for the compliance test. It indicates that codec set 2 will be used for calls in region 10 (the service provider region) region 1 (the enterprise side) and region 2 (the IP phones). Creating this table entry for ip-network-region 1 will automatically create a complementary table entry on the ip-network-region 1 and 2 forms for destination region 10. change ip-network-region 10 Page 3 of 19 Source Region: 10 Inter Network Region Connection Management I M G A t dst codec direct WAN-BW-limits Video Intervening Dyn A G c rgn set WAN Units Total Norm Prio Shr Regions CAC R L e 1 2 y NoLimit n t 2 2 y NoLimit n t all Signaling Group Use the add signaling-group command to create a signaling group between Communication Manager and the SBC for use by the service provider trunk. This signaling group is used for inbound and outbound calls between the service provider and the enterprise. For the compliance test, signaling group 2 was used for this purpose and was configured using the parameters highlighted below. Set the Group Type field to sip. Set the Transport Method to the recommended default value of tls (Transport Layer Security). For ease of troubleshooting during testing, part of the compliance test was conducted with the Transport Method set to tcp. The transport method specified here is used between the Communication Manager and Session Manager. Set the Near-end Listen Port and Far-end Listen Port to a valid unused port instead of the default well-known port value. (For TLS, the well-known port value is 5061 and for TCP the well-known port value is 5060). The compliance test was conducted with the Near-end Listen Port and Far-end Listen Port set to Set the Near-end Node Name to procr. This node-name maps to the IP address of the S8300C as defined in the node-names ip screen shot in Section of 75

14 Set the Far-end Node Name to ASM. This node name maps to the IP address of the Session Manger SM100 interface as defined in the node-names-ip screen shot in Section 4.3. Set the Far-end Network Region to the IP network region for the service provider in Section 4.5. Set the Far-end Domain to qwest.com. Communication Manager identifies incoming calls based on the P-Asserted Identity s host name, matching the value entered here. It is also used to populate the host field in the To header for outgoing calls. Set Direct IP-IP Audio Connections to y. This field will enable media shuffling on the SIP trunk allowing Communication Manager to redirect media traffic directly between the SIP trunk and the enterprise endpoint. Set the DTMF over IP field to rtp-payload. This value enables Communication Manager to send DTMF transmissions using RFC Set the Alternate Route Timer to 5. This defines the number of seconds the that Communication Manager will wait for a response (other than 100 Trying) to an outbound INVITE before selecting another route. If an alternate route is not defined, then the call is cancelled after this interval. Default values may be used for all other fields. change signaling-group 2 Page 1 of 1 SIGNALING GROUP Group Number: 2 IMS Enabled? n IP Video? n Group Type: sip Transport Method: tcp Near-end Node Name: procr Far-end Node Name: ASM Near-end Listen Port: 5070 Far-end Listen Port: 5070 Far-end Network Region: 10 Far-end Domain: qwest.com Bypass If IP Threshold Exceeded? n Incoming Dialog Loopbacks: eliminate RFC 3389 Comfort Noise? n DTMF over IP: rtp-payload Direct IP-IP Audio Connections? y Session Establishment Timer(min): 3 IP Audio Hairpinning? n Enable Layer 3 Test? n Direct IP-IP Early Media? n H.323 Station Outgoing Direct Media? n Alternate Route Timer(sec): 5 14 of 75

15 4.7. Trunk Group Use the add trunk-group command to create a trunk group for the signaling-group created in Section 4.6. For the compliance test, trunk group 2 was configured using the parameters highlighted below. Set the Group Type field to sip. Enter a descriptive name for the Group Name. Enter an available trunk access code (TAC) that is consistent with the existing dial plan in the TAC field. Set the Service Type field to public-ntwrk. Set the Signaling Group to the signaling group shown in the previous step. Set the Number of Members field to the number of trunk members in the SIP trunk group. This value determines how many simultaneous SIP calls can be supported by this trunk. Default values were used for all other fields. add trunk-group 2 Page 1 of 21 TRUNK GROUP Group Number: 2 Group Type: sip CDR Reports: y Group Name: Qwest COR: 1 TN: 1 TAC: *102 Direction: two-way Outgoing Display? n Dial Access? n Night Service: Queue Length: 0 Service Type: public-ntwrk Auth Code? n Signaling Group: 2 Number of Members: of 75

16 On Page 2, the Redirect On OPTIM Failure value is the amount of time (in milliseconds) that Communication Manager will wait for a response (other than 100 Trying) to a pending INVITE sent to an EC500 remote endpoint before selecting another route. If another route is not defined, then the call is cancelled after this interval. This time interval should be set to a value comparable to the Alternate Route Timer on the signaling group form described in Section 4.6. Verify that the Preferred Minimum Session Refresh Interval is set to a value acceptable to the service provider. This value defines the interval that re-invites must be sent to keep the active session alive. For the compliance test, the value of 600 seconds was used. add trunk-group 2 Page 2 of 21 Group Type: sip TRUNK PARAMETERS Unicode Name: auto Redirect On OPTIM Failure: 5000 SCCAN? n Digital Loss Group: 18 Preferred Minimum Session Refresh Interval(sec): 600 Disconnect Supervision - In? y Out? y XOIP Treatment: auto Delay Call Setup When Accessed Via IGAR? On Page 3, make sure the Numbering Format field is set to public. Set the Replace Restricted Numbers? and the Replace Unavailable Numbers? fields to y This will allow the CPN displayed on local endpoints to be replaced with the value set in Section 4.2, if the inbound call enabled CPN block. For outbound calls, these same settings request that CPN block be activated on the far-end destination if a local user requests CPN block on a particular call routed out this trunk. Default values were used for all other fields. add trunk-group 2 Page 3 of 21 TRUNK FEATURES ACA Assignment? n Measured: none Maintenance Tests? y Numbering Format: public UUI Treatment: service-provider Replace Restricted Numbers? y Replace Unavailable Numbers? y 16 of 75

17 On Page 4, set the Network Call Redirection field to n. Set the Send Diversion Header field to y. This field provides additional information to the network if the call has been redirected. This is needed to support call forwarding of inbound calls back to the PSTN and some Extension to Cellular (EC500) call scenarios. Set the Telephone Event Payload Type to 100, the value preferred by Qwest iq SIP Trunk. change trunk-group 2 Page 4 of 21 PROTOCOL VARIATIONS Mark Users as Phone? n Prepend '+' to Calling Number? n Send Transferring Party Information? n Network Call Redirection? n Send Diversion Header? y Support Request History? n Telephone Event Payload Type: 100 Convert 180 to 183 for Early Media? n Always Use re-invite for Display Updates? n Identity for Calling Party Display: P-Asserted-Identity Enable Q-SIP? n 4.8. Calling Party Information The calling party number is sent in the SIP From, Contact and PAI headers. Since public numbering was selected to define the format of this number (Section 4.7), use the change public-numbering command to create an entry for the DID range assigned. The DID numbers are provided by the SIP service provider. Each DID number is assigned to one enterprise internal extension. It is used to authenticate the caller. In the sample configuration, Extensions were created matching the last 4 digits of the DID block. There is also extension numbers starting with a 4 and 5 that were mapped to the lead DID number as shown in Figure 1. change public-unknown-numbering 0 Page 1 of 2 NUMBERING - PUBLIC/UNKNOWN FORMAT Total Ext Ext Trk CPN CPN Len Code Grp(s) Prefix Len Total Administered: Maximum Entries: of 75

18 4.9. Outbound Routing In these Application Notes, the Automatic Route Selection (ARS) feature is used to route outbound calls via the SIP trunk to the service provider. In the sample configuration, the single digit 9 is used as the ARS access code. Enterprise callers will dial 9 to reach an outside line. This common configuration is illustrated below with little elaboration. Use the change dialplan analysis command to define a dialed string beginning with 9 of length 1 as a feature access code (fac). change dialplan analysis Page 1 of 12 DIAL PLAN ANALYSIS TABLE Location: all Percent Full: 1 Dialed Total Call Dialed Total Call Dialed Total Call String Length Type String Length Type String Length Type 1 3 dac 2 4 ext 4 4 ext 5 4 ext 6 4 ext 7 4 ext 8 1 fac 9 1 fac * 4 dac # 4 fac Use the change feature-access-codes command to configure 9 as the Auto Route Selection (ARS) Access Code 1. change feature-access-codes Page 1 of 9 FEATURE ACCESS CODE (FAC) Abbreviated Dialing List1 Access Code: #110 Abbreviated Dialing List2 Access Code: #111 Abbreviated Dialing List3 Access Code: #112 Abbreviated Dial - Prgm Group List Access Code: #113 Announcement Access Code: #114 Answer Back Access Code: Attendant Access Code: Auto Alternate Routing (AAR) Access Code: 8 Auto Route Selection (ARS) - Access Code 1: 9 Access Code 2: Automatic Callback Activation: Deactivation: Call Forwarding Activation Busy/DA: All: Deactivation: Call Forwarding Enhanced Status: Act: Deactivation: Call Park Access Code: Call Pickup Access Code: CAS Remote Hold/Answer Hold-Unhold Access Code: CDR Account Code Access Code: 18 of 75

19 Use the change ars analysis command to configure the routing of dialed digits following the first digit 9. The example below shows a subset of the dialed strings tested as part of the compliance test. See Section 1.1 for the complete list of call types tested. All dialed strings are mapped to route pattern 2 which contains the SIP trunk to the service provider (as defined next). change ars analysis 1 Page 1 of 2 ARS DIGIT ANALYSIS TABLE Location: all Percent Full: 0 Dialed Total Route Call Node ANI String Min Max Pattern Type Num Reqd fnpa n fnpa n fnpa n fnpa n fnpa n fnpa n fnpa n fnpa n 1xxx deny fnpa n hnpa n svcl n hnpa n svcl n n 19 of 75

20 The route pattern defines which trunk group will be used for the call and performs any necessary digit manipulation. Use the change route-pattern command to configure the parameters for the service provider trunk route pattern in the following manner. The example below shows the values used for route pattern 2 during the compliance test. Pattern Name: Enter a descriptive name. Grp No: Enter the outbound trunk group for the SIP service provider. For the compliance test, trunk group 2 was used. FRL: Set the Facility Restriction Level (FRL) field to a level that allows access to this trunk for all users that require it. The value of 0 is the least restrictive level. Pfx Mrk: 1 The prefix mark (Pfx Mrk) of one will prefix any FNPA 10-digit number with a 1 and leave numbers of any other length unchanged. This will ensure digits are sent to the service provider for long distance North American Numbering Plan (NANP) numbers. All HNPA 10 digit numbers are left unchanged. change route-pattern 2 Page 1 of 3 Pattern Number: 2 Pattern Name: Outbound PSTN SCCAN? n Secure SIP? n Grp FRL NPA Pfx Hop Toll No. Inserted DCS/ IXC No Mrk Lmt List Del Digits QSIG Dgts Intw 1: n user 2: n user 3: n user 4: n user 5: n user 6: n user BCC VALUE TSC CA-TSC ITC BCIE Service/Feature PARM No. Numbering LAR M 4 W Request Dgts Format Subaddress 1: y y y y y n n rest none 2: y y y y y n n rest none 3: y y y y y n n rest none 20 of 75

21 Use the change ars digit-conversion command to manipulate the routing of dialed digits that match the DIDs to prevent the calls from going out the PSTN and routing to the proper extension. The example below shows the toll-free number being converted to a VDN and the DID block being converted to the extensions matching the last 4 digits. change ars digit-conversion 0 Page 1 of 2 ARS DIGIT CONVERSION TABLE Location: all Percent Full: 0 Matching Pattern Min Max Del Replacement String Net Conv ANI Req ext y n ext y n ext y n ext y n ext y n ext y n n n n n n n n Incoming Call Handling Treatment In general, the incoming call handling treatment for a trunk group can be used to manipulate the digits received for an incoming call if necessary. The toll-free number sent by Qwest iq SIP Trunk can be mapped to an extension using the incoming call handling treatment of the receiving trunk group. As an example, the following screen illustrates a conversion of toll-free number to extension It also shows the range of DID s being converted to 4 digit extensions. change inc-call-handling-trmt trunk-group 2 Page 1 of 3 INCOMING CALL HANDLING TREATMENT Service/ Number Number Del Insert Feature Len Digits public-ntwrk public-ntwrk public-ntwrk public-ntwrk public-ntwrk public-ntwrk public-ntwrk public-ntwrk public-ntwrk public-ntwrk public-ntwrk public-ntwrk public-ntwrk 21 of 75

22 5. Configure Avaya Aura Session Manager This section provides the procedures for configuring Session Manager. The procedures include adding the following items: SIP domain Logical/physical Location that can be occupied by SIP Entities Adaptation module to perform digit manipulation SIP Entities corresponding to Communication Manager, the AA-SBC and Session Manager Entity Links, which define the SIP trunk parameters used by Session Manager when routing calls to/from SIP Entities Routing Policies, which control call routing between the SIP Entities Dial Patterns, which govern to which SIP Entity a call is routed Regular Expressions, which also can be used to route calls Session Manager, corresponding to the Session Manager Server to be managed by System Manager. It may not be necessary to create all the items above when creating a connection to the service provider since some of these items would have already been defined as part of the initial Session Manager installation. This includes items such as certain SIP domains, locations, SIP entities, and Session Manager itself. However, each item should be reviewed to verify the configuration System Manager Login and Navigation Session Manager configuration is accomplished by accessing the browser-based GUI of System Manager, using the URL where <ip-address> is the IP address of System Manager. At the System Manager Log On screen, provide the appropriate credentials and click on Login On (not shown). The initial screen shown below is then displayed. Most of the configuration items are performed in the Routing Element. Click on Routing in the Elements column to bring up the Introduction to Network Routing Policy screen. 22 of 75

23 The navigation tree displayed in the left pane will be referenced in subsequent sections to navigate to items requiring configuration Specify SIP Domain Create a SIP domain for each domain for which Session Manager will need to be aware of in order to route calls. For the compliance test, this includes the enterprise domain (sip.avaya.com) and the Qwest iq SIP trunk domain (qwest.com). Navigate to Routing Domains in the left-hand navigation pane and click the New button in the right pane (not shown). In the new right pane that appears (shown below), fill in the following: Name: Type: Notes: Enter the domain name. Select sip from the pull-down menu. Add a brief description (optional). Click Commit. The screen below shows the entry for the Qwest domain. 23 of 75

24 The screen below shows both the enterprise domain and the Qwest domain after they have been created. 24 of 75

25 5.3. Add Location Locations can be used to identify logical and/or physical locations where SIP Entities reside for purposes of bandwidth management and call admission control. To add a location, navigate to Routing Locations in the left navigation pane and click the New button in the right pane (not shown). In the General section, enter the following values. Use default values for all remaining fields: Name: Enter a descriptive name for the location. Notes: Add a brief description (optional). In the Location Pattern section, click Add and enter the following values. Use default values for all remaining fields: IP Address Pattern: An IP address pattern used to identify the location. Notes: Add a brief description (optional). Displayed below are the top and bottom halves of the screen for addition of the Westminster Location, which includes all equipment on the x subnet including Communication Manager, the IP phones, and the Session Manager itself. Click Commit to save. Note: that call bandwidth management parameters should be set per customer requirement. 25 of 75

26 Repeat the preceding procedure to create a separate Location for AA-SBC. Displayed below is the screen for addition of the AA-SBC Location, which specifies the specific IP address for the AA-SBC. Click Commit to save Add Adaptation Module Session Manager can be configured with Adaptation modules that modify SIP messages before or after routing decisions have been made. A generic Adaptation module DigitConversionAdapter supports digit conversion of telephone numbers in specific headers of SIP messages. Other Adaptation modules are built on this generic module, and can modify other headers to permit interoperability with third party SIP products. For Qwest iq SIP Trunk interoperability, one Adaptation is needed and maps inbound DID numbers from Qwest SIP Trunk to local Communication Manager extensions. The adaptation will later on be applied to the AA-SBC SIP Entity. To create the adaptation, navigate to Routing Adaptations in the left navigation pane and click on the New button in the right pane (not shown). In the General section, enter the following values. Use default values for all remaining fields: Adaptation name: Enter a descriptive name for the adaptation. Module name: Enter DigitConversionAdapter. Module parameter: Enter iosrcd=qwest.com. This setting adapts the incoming call (i.e., from SBC to Session Manager) destination domain to the 26 of 75

27 domain expected by Communication Manager in the sample configuration. Defaults can be used for the remaining fields. Click Commit to save Add SIP Entities A SIP Entity must be added for Session Manager and for each SIP telephony system connected to it which includes Communication Manager and the AA-SBC. Navigate to Routing SIP Entities in the left navigation pane and click on the New button in the right pane (not shown). In the General section, enter the following values. Use default values for all remaining fields: Name: Enter a descriptive name. FQDN or IP Address: Enter the FQDN or IP address of the SIP Entity that is used for SIP signaling. Type: Enter Session Manager for Session Manager, CM for Communication Manager and SIP Trunk for the AA-SBC. Adaptation: This field is only present if Type is not set to Session Manager. If applicable, select the Adaptation Name created in Section 5.4 that will be applied to this entity. Location: Select one of the locations defined previously. Time Zone: Select the time zone for the location above. 27 of 75

28 The following screen shows the addition of Session Manager. The IP address of the virtual SM-100 Security Module (the Session Manager signaling interface) is entered for FQDN or IP Address. To define the ports used by Session Manager, scroll down to the Port section of the SIP Entity Details screen. This section is only present for the Session Manager SIP Entity. In the Port section, click Add and enter the following values. Use default values for all remaining fields: Port: Port number on which the Session Manager can listen for SIP requests. Protocol: Transport protocol to be used to send SIP requests. Default Domain: The domain used for the enterprise. Defaults can be used for the remaining fields. Click Commit to save. 28 of 75

29 The following screen shows the addition of Communication Manager. The FQDN or IP Address field is set to the IP address of the Processor Ethernet Interface of the Communication Manager S of 75

30 The following screen shows the addition of the AA-SBC SIP Entity. The FQDN or IP Address field is set to the IP address of its private network interface (see Figure 1). Link Monitoring Enabled was selected for SIP Link Monitoring using the specific time settings for Proactive Monitoring Interval (in seconds) and Reactive Monitoring Interval (in seconds) for the compliance test. These time settings should be adjusted or left at their default values per customer needs and requirements. For the Adaptation field, select the adaptation module previously defined for dial plan digit manipulation in Section Add Entity Links A SIP trunk between Session Manager and a telephony system is described by an Entity Link. Two Entity Links were created; one to the Communication Manager and one to the AA-SBC. To add an Entity Link, navigate to Routing Entity Links in the left navigation pane and click on the New button in the right pane (not shown). Fill in the following fields in the new row that is displayed: Name: SIP Entity 1: Protocol: Port: Enter a descriptive name. Select the Session Manager. Select the transport protocol used for this link. Port number on which Session Manager will receive SIP requests from the far-end. For the Communication Manager, this must match the Far-end Listen Port defined on the Communication Manager signaling group in Section of 75

31 SIP Entity 2: Port: Trusted: Select the name of the other system. For Communication Manager, select the Communication Manager SIP Entity defined in Section 5.5. For AA-SBC, select the AA-SBC SIP Entity defined in Section 5.5. Port number on which the other system receives SIP requests from the Session Manager. For the Communication Manager, this must match the Near-end Listen Port defined on the Communication Manager signaling group in Section 4.6. Check this box. Note: If this box is not checked, calls from the associated SIP Entity specified in Section 5.5 will be denied. Click Commit to save. The following screens illustrate the Entity Links to Communication Manager and the AA-SBC. It should be noted that in a customer environment the Entity Link to Communication Manager would normally use TLS. For the compliance test, TCP was used to aid in troubleshooting since the signaling traffic would not be encrypted. The protocol and ports defined here must match the values used on the Communication Manager signaling group form in Section 4.6 Entity Link to Communication Manager: Entity Link to the AA-SBC: 31 of 75

32 5.7. Add Routing Policies Routing policies describe the conditions under which calls will be routed to the SIP Entities specified in Section 5.5. Two routing policies must be added: one for Communication Manager and one for the AA-SBC. To add a routing policy, navigate to Routing Routing Policies in the left navigation pane and click on the New button in the right pane (not shown). The following screen is displayed. Fill in the following: In the General section, enter the following values. Use default values for all remaining fields: Name: Enter a descriptive name. Notes: Add a brief description (optional). In the SIP Entity as Destination section, click Select. The SIP Entity List page opens (not shown). Select the appropriate SIP entity to which this routing policy applies and click Select. The selected SIP Entity displays on the Routing Policy Details page as shown below. Use default values for remaining fields. Click Commit to save. The following screens show the Routing Policies for Communication Manager and the AA- SBC. 32 of 75

33 5.8. Add Dial Patterns Dial Patterns are needed to route calls through Session Manager. For the compliance test, dial patterns were needed to route calls from Communication Manager to Qwest and vice versa. Dial Patterns define which route policy will be selected for a particular call based on the dialed digits, destination domain and originating location. To add a dial pattern, navigate to Routing Dial Patterns in the left navigation pane and click on the New button in the right pane (not shown). Fill in the following, as shown in the screens below: In the General section, enter the following values. Use default values for all remaining fields: Pattern: Enter a dial string that will be matched against the Request-URI of the call. Min: Enter a minimum length used in the match criteria. Max: Enter a maximum length used in the match criteria. SIP Domain: Enter the destination domain used in the match criteria. Notes: Add a brief description (optional). In the Originating Locations and Routing Policies section, click Add. From the Originating Locations and Routing Policy List that appears (not shown), select the appropriate originating location for use in the match criteria. Lastly, select the routing policy from the list that will be used to route all calls that match the specified criteria. Click Select. Default values can be used for the remaining fields. Click Commit to save. Two examples of the dial patterns used for the compliance test are shown below, one for outbound calls from the enterprise to the PSTN and one for inbound calls from the PSTN to the enterprise. Other dial patterns (e.g., 011 international calls, 411 directory assistance calls, etc.), were similarly defined. 33 of 75

34 The first example shows that 11 digit dialed numbers that begin with 1 uses route policy To_AA-SBC01. The second example shows that inbound 10 digit numbers that start with and originating from Location AA-SBC uses route policy To_S8300c_CM521. These are the DID numbers assigned to the enterprise from Qwest. 34 of 75

35 The screen below shows a list of dial patterns used for the compliance test. 35 of 75

36 5.9. Add/View Session Manager The creation of a Session Manager element provides the linkage between System Manager and Session Manager. This was most likely done as part of the initial Session Manager installation. To add a Session Manager, navigate to Home Elements Session Manager Session Manager Administration in the left navigation pane and click on the New button in the right pane (not shown). If the Session Manager already exists, click View (not shown) to view the configuration. Enter/verify the data as described below and shown in the following screen: In the General section, enter the following values: SIP Entity Name: Description: Management Access Point Host Name/IP: Select the SIP Entity created for Session Manager. Add a brief description (optional). Enter the IP address of the Session Manager management interface. The screen below shows the Session Manager values used for the compliance test. 36 of 75

37 In the Security Module section, enter the following values: SIP Entity IP Address: Should be filled in automatically based on the SIP Entity Name. Otherwise, enter IP address of the Session Manager signaling interface. Network Mask: Enter the network mask corresponding to the IP address of Session Manager. Default Gateway: Enter the IP address of the default gateway for Session Manager. Use default values for the remaining fields. Click Save (not shown) to add this Session Manager. The screen below shows the remaining Session Manager values used for the compliance test. 37 of 75

38 6. Avaya Aura SBC Configuration 6.1. Initial Installation This section displays basic configuration of the SBC from the installation wizard. The initial configuration was completed by installing the Virtual Server Platform (VSP) and then logging into the web interface of the server (cdom) address. This screen will verify the state of the dom-0 and cdom platforms (the State column below) and the Application State of the SBC after it has been installed. After Installation of VSP is complete open the web interface to the CDOM and go to Virtual Machine Management Solution Template Select the location of the template: 38 of 75

39 Select the correct template for your server hardware: If an Electronic Preinstallation Worksheet has been created select Browse EPW File to select the location of the file then select Upload EPW file. Otherwise select Continue without EPW file. 39 of 75

40 Select Install after verifying: At this point, it will open a log that will show the progress of the installation. When it gets to Wait for User to Complete Data Entry, it will open another window for input and you may have to enable pop-ups to view. 40 of 75

41 During the installation of the Avaya Aura SBC template, the installation wizard will prompt the installer for information that will be used to create the initial configuration of the AA-SBC Network Settings The first screen of the installation wizard is the Network Settings screen. Fill in the fields as described below and shown in the following screen: IP Address: Enter the IP address of the private side of the AA-SBC. Hostname: Enter a host name for the AA-SBC. Domain: Enter the domain used for the enterprise. Default Domain: Enter the domain used for the enterprise. Click Next Step to continue. Note: In this version the domain that is listed is NOT optional and must be populated or your install will fail, please refer to the release notes for further information. 41 of 75

42 Logins Assign passwords for the different accounts. Click Next Step to continue VPN Access VPN remote access to the AA-SBC was not part of the compliance test. Thus, on the VPN Access screen, select No to the question, Would you like to configure the VPN remote access parameters for System Platform? Click Next Step to continue. 42 of 75

43 SBC On the SBC screen, fill in the fields as described below and shown in the following screen: In the SIP Service Provider Data section: Service Provider: From the pull-down menu, select the name of the service provider to which the AA-SBC will connect. This will allow the wizard to select a configuration file customized for this service provider. At the time of the compliance test, a customized configuration file did not exist for Qwest. Thus, Generic was chosen instead and further customization was done manually after the wizard was complete. Port: Enter the port number that the service provider uses to listen for SIP traffic. IP Address1: Enter the Qwest provided IP address of the Qwest SIP Proxy. Signaling/Media Network1: Signaling/Media Netmask1: IP Address2 (Optional): Enter the Qwest provided subnet where signaling/media traffic will originate. Enter the network mask corresponding to Signaling/Media Network 1. If an additional network has been provided, enter the second Qwest provided IP address of the Qwest SIP Proxy. Any additional networks can be added after installation 43 of 75

44 Signaling/Media Network2 (Optional): Signaling/Media Netmask2 (Optional): If an additional network is required, enter the second Qwest provided subnet where signaling/media traffic will originate. Any additional networks can be added after installation. Enter the network mask corresponding to Signaling/Media Network 2. In the SBC Network Data section: Public IP Address: Enter the IP address of the public side of the AA-SBC. Public Net Mask: Enter the netmask associated with the public network to which the SBC connects. Public Gateway: Enter the default gateway of the public network. In the Enterprise SIP Server section: SIP Domain: Enter the enterprise SIP domain. IP Address: Enter the IP address of the Enterprise SIP Server to which the AA-SBC will connect. In the case of the compliance test, this is the IP address of the Session Manager SIP signaling interface. Transport: From the pull-down menu, select the transport protocol to be used for SIP traffic between the AA-SBC and Session Manager. Click Next Step to continue. 44 of 75

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46 A summary screen will be displayed. Check the displayed values and click Next Step again to continue to the final step. 46 of 75

47 Confirm Installation The Confirm Installation screen will indicate if any required or optional fields have not been set. The list of required fields that have not been set should be empty. If not, click Previous Step to navigate to the relevant screen to set the required fields. Otherwise click Accept then Install to continue the overall template installation. 47 of 75

48 Wait until the template installation says Template Installation Completed Successfully. 48 of 75

49 Verify Installation Click on Home screen to verify the SBC virtual machine is running. Verify the proper IP interface is assigned to sbcpublic. Navigate to Server Management Network Configuration. 49 of 75

50 Scroll down to Domain Network Interface. Make sure the sbcpublic Interface is the same as the Template Network Configuration SBC Public Interface, eth2. Please refer to SBC installation manuals and appropriate release notes for further assistance. Also note, this version of SBC is the first to require the license file be loaded via weblm instead of the SBC web interface. 50 of 75

51 6.2. Post Installation Configuration The installation wizard configures the Session Border Controller for use with the service provider chosen in Section 6.1. Since the Generic service provider was selected in the installation wizard, additional manual changes need to be performed. These changes are performed by accessing the browser-based GUI of the AA-SBC, using the URL where <ip-address> is the private IP address configured in Section 6.1. Log in with the proper credentials Add Additional SIP Gateways Depending on the type of service, Qwest iq SIP Trunk may have multiple servers for redundancy purposes and remote DID s. During compliancy testing, three servers were used. Two servers were used for inbound/outbound dialing and one was used for inbound remote DID s and toll-free numbers. During the installation wizard the two used for inbound/outbound dialing were added. However, the two servers were in separate locations with separate subnets. When adding a new server that is in a separate subnet, it is necessary to add a route and a kernel-filter for it. To add a route, navigate to cluster box:<server_name> interface eth2 ip outside routing. Click on Add route. 51 of 75

52 Fill in the fields as described below and shown in the following screen: route-name: Enter a descriptive name. In the example below external-sipmedia-2 was used. destination: For type choose network. For address/mask enter the network and subnet mask provided by Qwest. gateway: Enter the IP address of the router connected to interface eth2. Click Set to complete the configuration. 52 of 75

53 In order for the AA-SBC to accept traffic from the new server a Kernel Filter needs to be added. Navigate to cluster box:<server_name> interface eth2 ip outside kernel-filter and click on Add allow-rule. Fill in the fields as described below and shown in the following screen: name: Enter a descriptive name. In the example below allow-sipudp-from-peer-2 was used. admin: Make sure it is set to enabled. 53 of 75

54 Destination-port: Enter the port number provided by Qwest. Typically source-address/mask: Enter the network and subnet mask provided by Qwest. source-port: Enter 0. protocol: Choose the protocol provided by Qwest. Typically UDP. Click Set to complete the configuration. The installation wizard only allows two servers to be added from Qwest so any additional servers will have to be added manually. To add a server, navigate to vsp enterprise servers. Click on Add sip-gateway. 54 of 75

55 In the general section, enter the following values. Use default values for all remaining fields: name: Enter a descriptive name. In the example below TelecoDID was used. admin: Make sure it is set to enabled. domain: Leave blank. Failover-detection: Select ping. This sends SIP OPTIONS to detect failures. In the policy section, set the outbound-session-config-pool-entry to vsp\session-configpool\entry ToTelco. 55 of 75

56 Click Set to complete the configuration. 56 of 75

57 Next, in the servers section click on configure next to sever-pool. In the new right pane that appears click on Add Server. 57 of 75

58 In the general section, enter the following values: server name: Enter a descriptive name. In the example below TelecoDID1 was used. host: Enter the IP address provided by Qwest. Click on Create. 58 of 75

59 Next make sure the transport and port match what Qwest has provided. In this example the transport is set to UDP and the port is set to Leave everything else default. Click Set to complete the configuration. 59 of 75

60 A dial plan needs to be created to route calls from the new Qwest inbound server to the Session Manager. Sense the new server is for inbound only; a route to the server from Session Manager is not needed. Navigate to vsp dial-plan. Click on Add source-route. In the general section, enter the following values: name: Enter a descriptive name. In the example below FromTelecoDID was used. Source-match: Set type to server and source-server to vsp\enterprise\servers\sip-gateway TelcoDID. Click on Create. 60 of 75

61 In the general section, enter the following values: peer: Set type to server and server to vsp\enterprise\servers\sipgateway PBX. Leave everything else default. Click Set to complete the configuration. 61 of 75

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63 Blocked Headers The P-Location and P-Charging-Vector headers are sent in SIP messages from the Session Manager. These headers should not be exposed external to the service provider. For simplicity, these headers were simply removed (blocked) from both requests and responses for both inbound and outbound calls. To create a rule for blocking a header on an outbound call, first navigate to vsp default-session-config header-settings. Click Edit blocked-header. 63 of 75

64 In the right pane that appears, click Add. In the blank field that appears, enter the name of the header to be blocked. After all the blocked headers are added, click OK. The screen below shows the P-Location header and the P-Charging-Vector header were configured to be blocked for the compliance test. Click OK to continue. The list of blocked headers will appear in the right pane as shown below. Click Set to complete the configuration. 64 of 75

65 Third Party Call Control Disable third party call control. Navigate to vsp default-session-config third-partycall-control. Set the admin field to disabled. Click Set to complete the configuration Save the Configuration To save the configuration, begin by clicking on Configuration in the left pane to display the configuration menu. Next, select Update and save configuration. 65 of 75